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You can absolutely hear the difference between 44.1 and 96 kHz sample rate. Even with typical reproduction filters on the output, sampling at 44.1 kHz prevents

You can absolutely hear the difference between 44.1 and 96 kHz sample rate. Even with typical reproduction filters on the output, sampling at 44.1 kHz prevents you from accurately preserving e.g. a sine tone at above ~6 kHz, and that's even without taking into account all the aliasing problems you're facing when the samples don't align with with the peaks of this or that tone. 44.1 kHz is "good enough", but it's not accurate, and you can definitely tell the difference. As for anything beyond 16 bits amplitude on line level, no, you cannot hear a difference. For such a low-voltage signal the resolution at 16 bits is so fine that it already drowns in all the natural noise and THD in the cables, in the amplifier, in your speakers/headphones etc.

pja 3y ago
Please do watch & internally digest the explanatory videos at https://xiph.org/video/ https://xiph.org/video/ It explains why you’re wrong in easily digestible terms & how a 44kHz sample rate will accurately encode signals right up to the Nyquist limit. The second video is an end to end demo showing the process in action.
- user_7832 3y ago
  > https://xiph.org/video/ https://xiph.org/video/ Thanks a lot for those videos, they were absolutely excellent. For anyone wondering they're presented by "Monty", the guy behind the ogg container and vorbis codec. I probably understood 10% of what he said but that's still a lot.
temp0826 3y ago
It's been a long time since my DSP classes at uni, but I don't think this is true. 44.1kHz sampling is enough to reproduce up to 22.05kHz sound accurately without aliasing. Unless there is another type of distortion you might be picking up. This stuff is pretty far out of my realm these days. https://en.wikipedia.org/wiki/Nyquist_frequency https://en.wikipedia.org/wiki/Nyquist_frequency
- daneel_w 3y ago
  It's true for a triangle wave and a square wave, depending on if the output has a reconstruction filter doing linear interpolations between samples. You cannot accurately sample a 22.05 kHz sine wave (or any other "complex" waveform) with a 44.1 kHz sample rate.
  - vel0city 3y ago
    You're incorrect. A 44.1kHz sample rate can recreate a 22.05kHz sine wave. Your mental model for a digital signal isn't correct. It's not making stair steps nor is it making triangular waves. https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB
vel0city 3y ago
> sampling at 44.1 kHz prevents you from accurately preserving e.g. a sine tone at above ~6 kHz This is mathematically false. A 6kHz or 8kHz or 10kHz or 20kHz signal absolutely can be perfectly preserved with a 44.1kHz sample rate. Not just kind of preserved, but perfectly preserved.
- masfuerte 3y ago
  It's perfectly preserved only if your samples are perfect. Imagine instead that we used 4-bit samples. The results would be obviously garbage. 8-bit would be better. 16-bit is better still. But it isn't perfect.
  - vel0city 3y ago
    Increasing the bitrate lowers the noise floor. But that 6 or 8kHz signal is still there. 8:43 in the second video, he goes in to showing what increasing the bit depth gets you. https://xiph.org/video/vid2.shtml https://xiph.org/video/vid2.shtml
- daneel_w 3y ago
  I doesn't look like you understand what sampling is, and how reconstruction filters in DACs work. Your statement is true for some waveforms, depending on their frequency, due to the use of reconstruction filters on the output, but it's not true for any signal and the problem becomes more apparent the higher the frequency of the waveform.
  - deleted 3y ago
    [deleted]
  - krackers 3y ago
    If I'm understanding you correctly, you're saying that while a perfect sinc interpolation reconstruction would allow you to capture up to 44.1/2 kHz, in practice since we're limited to FIR reconstruction filters we can't actually get that high? If so it seems like a fair point, although I'd imagine they'd be better than 6khz? There's also the issue of the input signal not being band-limited which is necessarily true for real world signals given that you sample for a finite duration.
ReactiveJelly 3y ago
Yeah this is the "stairstep vs. lollipop" thing again.
- kevin_thibedeau 3y ago
  Everyone complaining with "but stairstepping" fails to recognize that the final stage of a DAC is a reconstruction filter. The steps are gone after that filter is applied. You aren't analyzing the full DAC performance if you look in front of the filter. This is most dramatic in class-D amplifiers where the raw waveform feeding into the speakers is square wave hash that gets filtered out by the speakers themselves.
  - daneel_w 3y ago
    The filters do linear interpolation between samples. This bridges some shortcomings of a sample rate too low to capture complex waveform at high frequency, but it's not a silver bullet.
    - baobrien 3y ago
      They don't. Analog low-pass filters don't do linear interpolation between samples.
- daneel_w 3y ago
  Yes and no. Reconstruction filters are part of the problem (and part of the solution) but it's not all about them.
eyegor 3y ago
You should watch this: https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB to better understand why high sample rates are a waste of time, instead of just reading about nyquist. "accurately preserving a 6kHz sine wave" sounds a lot like you think that sample points are reproduced 1-1 from the digital to the analog domain. This just builds on the xiph video someone else linked but essentially - sine waves are fine as long as you have points for rising and falling edge (nyquist, 44k guarantees 22k sine wave reproduction) - bit depth only really affects noise floor, so it depends on your audios dynamic range
- daneel_w 3y ago
  A 44 kHz sample rate guarantees accurate 22 kHz triangle wave reproduction if a reconstruction filter with linear interpolation is used on the output, and accurate amplitude of same signal if samples happen to align somewhat with the peaks of the waveform.
  - NobodyNada 3y ago
    That's exactly why linear interpolation is not used. The Nyquist-Shannon sampling theorem says that you can perfectly reconstruct any sampled signal as long as the input signal was bandlimited to 1/2 the sampling rate. Using a low-pass filter to bandlimit your 22 kHz triangle wave will remove all the (inaudible) overtones, leaving you with a single 22 kHz sine wave as input to your ADC. The reconstruction filter on the output DAC will then output a perfect 22 kHz sine wave, with the correct amplitude too!
  - user_7832 3y ago
    Yep, as Monty showed in the 2nd xiph video a square wave will have issues with a low nquist frequency (at for eg 44khz sampling).
    - NobodyNada 3y ago
      There are no "issues" with the square wave at 44kHz. Yes, a square wave sampled at 44kHz and converted back to analog does have visible ringing on an oscilloscope, but that's because you're looking at a perfect reconstruction of a band-limited square wave. Before sampling, the square wave is filtered to remove all of the overtones above 22kHz, which are not audible (as that's above the upper limit of human hearing) and would cause aliasing issues in the sampled signal. In other words, the ringing is an artifact of low-pass-filtering on the square wave, not the sampling process itself. A purely analog system with a 22kHz low-pass filter would have the same "ringing", and your ears couldn't tell either way because they can't detect information above 15-20 kHz. It's not even possible to have a "perfect" square wave in the real world because that would require the speaker cone, air molecules, and your eardrum to teleport instantly from one spot to another -- so you will always have some low-pass filtering, and therefore ringing, on any real-world approximation of a square wave.

pja 3y ago

Please do watch & internally digest the explanatory videos at https://xiph.org/video/ https://xiph.org/video/ It explains why you’re wrong in easily digestible terms & how a 44kHz sample rate will accurately encode signals right up to the Nyquist limit. The second video is an end to end demo showing the process in action.

user_7832 3y ago
> https://xiph.org/video/ https://xiph.org/video/ Thanks a lot for those videos, they were absolutely excellent. For anyone wondering they're presented by "Monty", the guy behind the ogg container and vorbis codec. I probably understood 10% of what he said but that's still a lot.

temp0826 3y ago

It's been a long time since my DSP classes at uni, but I don't think this is true. 44.1kHz sampling is enough to reproduce up to 22.05kHz sound accurately without aliasing. Unless there is another type of distortion you might be picking up. This stuff is pretty far out of my realm these days. https://en.wikipedia.org/wiki/Nyquist_frequency https://en.wikipedia.org/wiki/Nyquist_frequency

daneel_w 3y ago
It's true for a triangle wave and a square wave, depending on if the output has a reconstruction filter doing linear interpolations between samples. You cannot accurately sample a 22.05 kHz sine wave (or any other "complex" waveform) with a 44.1 kHz sample rate.
- vel0city 3y ago
  You're incorrect. A 44.1kHz sample rate can recreate a 22.05kHz sine wave. Your mental model for a digital signal isn't correct. It's not making stair steps nor is it making triangular waves. https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB

vel0city 3y ago

> sampling at 44.1 kHz prevents you from accurately preserving e.g. a sine tone at above ~6 kHz This is mathematically false. A 6kHz or 8kHz or 10kHz or 20kHz signal absolutely can be perfectly preserved with a 44.1kHz sample rate. Not just kind of preserved, but perfectly preserved.

masfuerte 3y ago
It's perfectly preserved only if your samples are perfect. Imagine instead that we used 4-bit samples. The results would be obviously garbage. 8-bit would be better. 16-bit is better still. But it isn't perfect.
- vel0city 3y ago
  Increasing the bitrate lowers the noise floor. But that 6 or 8kHz signal is still there. 8:43 in the second video, he goes in to showing what increasing the bit depth gets you. https://xiph.org/video/vid2.shtml https://xiph.org/video/vid2.shtml
daneel_w 3y ago
I doesn't look like you understand what sampling is, and how reconstruction filters in DACs work. Your statement is true for some waveforms, depending on their frequency, due to the use of reconstruction filters on the output, but it's not true for any signal and the problem becomes more apparent the higher the frequency of the waveform.
- deleted 3y ago
  [deleted]
- krackers 3y ago
  If I'm understanding you correctly, you're saying that while a perfect sinc interpolation reconstruction would allow you to capture up to 44.1/2 kHz, in practice since we're limited to FIR reconstruction filters we can't actually get that high? If so it seems like a fair point, although I'd imagine they'd be better than 6khz? There's also the issue of the input signal not being band-limited which is necessarily true for real world signals given that you sample for a finite duration.

ReactiveJelly 3y ago

Yeah this is the "stairstep vs. lollipop" thing again.

kevin_thibedeau 3y ago
Everyone complaining with "but stairstepping" fails to recognize that the final stage of a DAC is a reconstruction filter. The steps are gone after that filter is applied. You aren't analyzing the full DAC performance if you look in front of the filter. This is most dramatic in class-D amplifiers where the raw waveform feeding into the speakers is square wave hash that gets filtered out by the speakers themselves.
- daneel_w 3y ago
  The filters do linear interpolation between samples. This bridges some shortcomings of a sample rate too low to capture complex waveform at high frequency, but it's not a silver bullet.
  - baobrien 3y ago
    They don't. Analog low-pass filters don't do linear interpolation between samples.
daneel_w 3y ago
Yes and no. Reconstruction filters are part of the problem (and part of the solution) but it's not all about them.

eyegor 3y ago

You should watch this: https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB https://youtu.be/cD7YFUYLpDc?si=rUm6IR3IKXyzcaDB to better understand why high sample rates are a waste of time, instead of just reading about nyquist. "accurately preserving a 6kHz sine wave" sounds a lot like you think that sample points are reproduced 1-1 from the digital to the analog domain. This just builds on the xiph video someone else linked but essentially - sine waves are fine as long as you have points for rising and falling edge (nyquist, 44k guarantees 22k sine wave reproduction) - bit depth only really affects noise floor, so it depends on your audios dynamic range

daneel_w 3y ago
A 44 kHz sample rate guarantees accurate 22 kHz triangle wave reproduction if a reconstruction filter with linear interpolation is used on the output, and accurate amplitude of same signal if samples happen to align somewhat with the peaks of the waveform.
- NobodyNada 3y ago
  That's exactly why linear interpolation is not used. The Nyquist-Shannon sampling theorem says that you can perfectly reconstruct any sampled signal as long as the input signal was bandlimited to 1/2 the sampling rate. Using a low-pass filter to bandlimit your 22 kHz triangle wave will remove all the (inaudible) overtones, leaving you with a single 22 kHz sine wave as input to your ADC. The reconstruction filter on the output DAC will then output a perfect 22 kHz sine wave, with the correct amplitude too!
- user_7832 3y ago
  Yep, as Monty showed in the 2nd xiph video a square wave will have issues with a low nquist frequency (at for eg 44khz sampling).
  - NobodyNada 3y ago
    There are no "issues" with the square wave at 44kHz. Yes, a square wave sampled at 44kHz and converted back to analog does have visible ringing on an oscilloscope, but that's because you're looking at a perfect reconstruction of a band-limited square wave. Before sampling, the square wave is filtered to remove all of the overtones above 22kHz, which are not audible (as that's above the upper limit of human hearing) and would cause aliasing issues in the sampled signal. In other words, the ringing is an artifact of low-pass-filtering on the square wave, not the sampling process itself. A purely analog system with a 22kHz low-pass filter would have the same "ringing", and your ears couldn't tell either way because they can't detect information above 15-20 kHz. It's not even possible to have a "perfect" square wave in the real world because that would require the speaker cone, air molecules, and your eardrum to teleport instantly from one spot to another -- so you will always have some low-pass filtering, and therefore ringing, on any real-world approximation of a square wave.